Hi, logic (as an application, running on any current mac with appropriate I/Os, drives etc) is easily able to do this. I routinely record 16-24 24/48khz tracks with 10 mono outputs (1x stereo control room, 4xstereo fold-back sub mixes) with nominal CPU usage and 10-20% HD meter usage. I don't use input monitoring as I do this in my interface (I love my MIOs!) but I can't imagine this would add much CPU load. I've recently started recording at 96Khz (24 bit) and this does show considerably more CPU and disk I/o usage but I'm still only at 20-40% CPU and 30-40% hard drive. I'm liberal with my usage of plugins and software instruments at mix down and only face issues with vast numbers of plugin saturated audio tracks and particularly hungry software instruments (I'm talking 80+ audio tracks/16+ instruments a couple of dozen assorted buses/auxes etc) at which point I have to start freezing stuff. There are obvious caveats here regarding disk drives/CPU(s)/RAM but I could do 16in/8out on my venerable G4 PowerBook 10 years ago! (Logic 6/7?)

I'm pretty confident they'll both work, I've just not had the need/equipment to try it! The only potential issues would tend to be hardware based (FW/TB/USB bus capacities to transfer the data to/from interface(s) and HD(s)) but these are the same with any DAW.

The only potential issues would tend to be hardware based (FW/TB/USB bus capacities to transfer the data to/from interface(s) and HD(s)) but these are the same with any DAW.

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Agreed. The issue here is likely to be the hardware audio connection bandwidths, drive speeds and so on. Not Logic, or any other DAW for that matter. I wonder if your friend's problem is hardware, rather than PT.

Interesting Pete -but he has not made a direct comparison with what I am doing. He is talking about CPU load. That part of it is not in doubt. His foldback for the active instruments comes off his mic pres. It is not passed through the CPU with any plugins on. All my 12 foldbacks come completely from PT with the benefit of all the plugs I use to get the sound. The Foldback mix is my running mix, except for the actual foldback mix track levels. I specialise in whole band drop-ins so 27 tracks up from the floor of the studio with all the mix plugs in playing back the tracks to the cans and then a quick punch in on all the tracks. Never any latency until I put a native plugin in. Even the non TDM instrument plugs cause latency on my TDM system.There are many workarounds to do this on a native system but I donâ€™t have to do any workarounds. For anyone not specialising in recording groups live on the floor and who doesnâ€™t put all the mix plugs in while tracking, native will work better because there is no doubt that my TDM system is no longer friends with my 5,1 Mac Pro as you have witnessed.
jxx

Foldback in this case means that the incoming recording signal goes to the computer, to Logic, through all the plugins and out to the musicians? This will always produce latency, on any system. The plugins need time to process the sound and this time can be quite remarkable depending on the type of plugins.

The only way I know of is to use an interface or recording mixer with built-in DSP and use it's effects for the foldback.

Maybe he's got the buffers so small that no one notices the latency until certain plugins are inserted, but it's obviously there, even without plugins, if an audio interface and DAW is being used. Again, it appears that the issue is not the DAW.

the whole point of TDM(or the new Avid version HDX) is that the real time processing is not done on the computer. The CPU draws the pictures on the screen and handles the disk I/O. Thats it. All the A-D/D-A processing and all the realtime application of plug-ins is done by the Digi interfaces and the 5 PCIe cards in the chassis. The only connection to the CPU is for drawing the pictures. The monitoring is not from the CPU either. So there is not as much load on the CPU as there is when you move large pics around on the screen in Photoshop. Think of TDM(Time Domain Multiplexing?) as the same as the parallel midi interface which was invented by Roland but never adopted - too expensive. We got stuck with a serial interface called midi. The equivalent is the native system where everything queues up to take its turn through the pipe - that causes the latency. But for most purposes the workarounds work very well.
jxx

To J -- yes, I understand. So you are not monitoring through the DAW. It's through hardware, right? But latency is happening, so either there's a problem with the hardware, or somehow the CPU is getting involved via software, and you are monitoring through the DAW. Other DAWs are not going to fix either of those issues.

I expect that with a reasonable computer, Logic, DP, Cubase, Sonar, etc., etc, would easily deal with 32 or more channels and 10 outputs with some effects at the same time. But not without latency unless the effects are hardware. But again, that has nothing to do with the capacity of the DAW.

The latency of the playback mix does not matter. Even if the playback comes with an offset of a couple of seconds, the musicians can only play along of what they hear and the latency compensation of the DAW puts the recorded data into the right location.

The latency plays only a role for the foldback mix, because there the musicians hear their own signals too late. So the only thing to work on is the monitor mix. Keep it as direct as possible. Best is splitting before the audio interface, or in the interface if the splitting happens in the analog domain. Second best is pseudo-direct monitoring through the audio interface (digital domain). Worst is monitoring through the DAW.

I have used software monitoring thru DAW on my latest album recording.
12 tracks live to disk with my RME FirefaceUC and 2.4GhZ Core2Duo MBPro.
We used no effects and just a reverb.
5 separate headphone mix.
No Problems about latency with 64K buffer.
It was a jazz recording so it had to be live.